[asterisk-users] Music on Hold

[Dan Saul]

That was a good shot in the dark, but sadly renaming it to something simple
(and removing all non ascii in the process) does not correct this.

On Wed, Sep 16, 2009 at 4:37 PM, Danny Nicholas <danny at debsinc.com> wrote:

>  Just a “shot in the dark” but could MOH be choking on the “long file
> names”?  (does it work on fred_chopin_pol_1)?
>
>
>  ------------------------------
>
> *From:* asterisk-users-bounces at lists.digium.com [mailto:
> asterisk-users-bounces at lists.digium.com] *On Behalf Of *Dan Saul
> *Sent:* Wednesday, September 16, 2009 4:18 PM
> *To:* asterisk-users at lists.digium.com
> *Subject:* [asterisk-users] Music on Hold
>
>
>
> Hi,
>
> I have trouble getting MOH to work after an upgrade from asterisk 1.4 to
> 1.6.1.4. The call goes on hold, MOH is started, and then stops right away.
>
> Here are the files both of type .raw:
>
> Tsunami*CLI> moh show files
> Class: default
>     File: /etc/asterisk/musiconhold/Frédéric Chopin - Polonaises Op. 40-2
>     File: /etc/asterisk/musiconhold/Frédéric Chopin - Polonaises Op. 40-1
>
> These files were generated by SoX:
> Channels       : 1
> Sample Rate    : 8000
> Precision      : 16-bit
> Sample Encoding: 16-bit Signed Integer PCM
> Endian Type    : little
> Reverse Nibbles: no
> Reverse Bits   : no
> Comment        : 'Processed by SoX'
>
> This prints in the asterisk console when you attempt to put someone in
> hold:
>
>     -- Started music on hold, class 'default', on
> SIP/link2voip-sw1-02477668
>     -- Stopped music on hold on SIP/link2voip-sw1-02477668
>
> No errors are printed, however the other side just hears silence.
>
> Here is the full debug output (asterisk -rvvvvv):
>
>  == Using SIP RTP CoS mark 5
>     -- Executing [xxxxxxx at phones:1] Goto("SIP/ATA-xxxxxxxxxx-L1-024b6d88",
> "1xxxxxxxxxx,1") in new stack
>     -- Goto (phones,1xxxxxxxxxx,1)
>     -- Executing [1xxxxxxxxxx at phones:1]
> MSet("SIP/ATA-xxxxxxxxxx-L1-024b6d88", "oldcidnum=0") in new stack
>     -- Executing [1xxxxxxxxxx at phones:2]
> MSet("SIP/ATA-xxxxxxxxxx-L1-024b6d88", "CALLERID(name)=""") in new stack
>     -- Executing [1xxxxxxxxxx at phones:3]
> MSet("SIP/ATA-xxxxxxxxxx-L1-024b6d88", "CALLERID(num)=xxxxxxxxxx") in new
> stack
>     -- Executing [1xxxxxxxxxx at phones:4]
> Monitor("SIP/ATA-xxxxxxxxxx-L1-024b6d88", "wav,/tmp/out 0 2009-09-17 03h 04m
> 51s CST xxxxxxxxxx,m") in new stack
>     -- Executing [1xxxxxxxxxx at phones:5]
> Gosub("SIP/ATA-xxxxxxxxxx-L1-024b6d88", "ExternalDial,s,1(1xxxxxxxxxx)") in
> new stack
>     -- Executing [s at ExternalDial:1] MSet("SIP/ATA-xxxxxxxxxx-L1-024b6d88",
> "LOCAL(num)=1xxxxxxxxxx") in new stack
>     -- Executing [s at ExternalDial:2] MSet("SIP/ATA-xxxxxxxxxx-L1-024b6d88",
> "~~EXTEN~~=s") in new stack
>     -- Executing [s at ExternalDial:3] Dial("SIP/ATA-xxxxxxxxxx-L1-024b6d88",
> "SIP/1xxxxxxxxxx at link2voip-sw1,120") in new stack
>   == Using SIP RTP CoS mark 5
>     -- Called 1xxxxxxxxxx at link2voip-sw1
>     -- SIP/link2voip-sw1-02477668 is making progress passing it to
> SIP/ATA-xxxxxxxxxx-L1-024b6d88
>     -- SIP/link2voip-sw1-02477668 answered SIP/ATA-xxxxxxxxxx-L1-024b6d88
>     -- Started music on hold, class 'default', on
> SIP/link2voip-sw1-02477668
>     -- Stopped music on hold on SIP/link2voip-sw1-02477668
>        > doing dnsmgr_lookup for 'sip.ca2.link2voip.com'
>        > doing dnsmgr_lookup for 'sip.ca1.link2voip.com'
>   == Spawn extension (ExternalDial, s, 3) exited non-zero on
> 'SIP/ATA-xxxxxxxxxx-L1-024b6d88'
>
> Any thoughts or ideas? If there were an error I could work on solving that,
> but there is none... Thanks.
>
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